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Hey everybody, I have a situation where we have 2 Asterisk (CVS as of 08/25/2005) connected via IAX. On the corporate side, we have 1 TE110P connecting to a Definity G3R and it's connecting to a TN464F card, giving a 23 channel connection. I have echocancel=yes, echotraining=yes and echocancelwhenbridged=yes. One the remote office side, they a Adit 600 channel bank for 10 outside

Can anyone help me here? ----- Forwarded message from Larry Rosenman <ler at lerctr.org> ----- Date: Sat, 9 Nov 2013 08:46:26 -0600 From: Larry Rosenman <ler at lerctr.org> To: freebsd-stable at freebsd.org Subject: 10-BETA3: Bad negotiation on AHD controller User-Agent: Mutt/1.5.22 (2013-10-16) Ever since I put 10 on this box (source upgrade from 8), I've been getting slow disk negotiations on 2 of my 4 disks, and weird messages at boot from the ahd controller. Can someone tell if it's my disk or the code? # for i in 0 1 2 3 >...

Is it not for a card with 4 FXO? I spent several hours the other day trying to figure out what I had done wrong and I ahd forgotten to connect the power cable. I setup several of these before and couldn't figure out why this one didn't work. It appears that's all it waqs. Without the power connecter the card will probe, and even appear to be working but when the lines ring (coming into the FXO port) it...

Hello, I'm a bit new to SIP, and I've set up a SIP line with Asterisk and my wholesale provider. That worked, fine. I ahd to open up the ports on my router, forward them to the correct box, again fine. Now, if I get one of my customers to connect his SIP phone to my Asterisk box, and HE'S behind a NAT firewall, does he have to go through the same process, or is it just the Asterisk box that needs to translate the...

...ase read in voip-info.org about SIP NAT, there are good suggestions. regards On 1/20/06, Michakl Gaudette <michael.gaudette@virtutel.ca> wrote: > Hello, > > I'm a bit new to SIP, and I've set up a SIP line with Asterisk and my > wholesale provider. That worked, fine. I ahd to open up the ports on my > router, forward them to the correct box, again fine. > > Now, if I get one of my customers to connect his SIP phone to my Asterisk > box, and HE'S behind a NAT firewall, does he have to go through the same > process, or is it just the Asterisk box tha...

In conjunction with my last post on Tellme I want to write another suggestion for an application I had. I don't know if you guys have come across Google Gas http://www.ahding.com/cheapgas But basically it is an application that this guy has developed using the Google API to search an online database on gas prices in your area. One of my strong beliefs about how Asterisk is going to leave the "Commercial" IP-PBX vendors behind is by leveraging the...

...out of the system .. which means we will need a call recroding.. will the four Asterisk servers handle the recording process or we will need external assistant? and if it was the second choice what is the best suggestion? is there a way to force an Asterisk server to record remote channels? -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 _________________________________________________________________ Express your personality in color! Preview and select themes for Hotmail?. http://www.windowslive-hotmail.com/LearnMore/pers...

...To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Is it mandatory to give power > supply to TDM400Pcard > > > Is it not for a card with 4 FXO? I spent several hours the > other day trying to figure out what I had done wrong and I > ahd forgotten to connect the power cable. > > I setup several of these before and couldn't figure out why > this one didn't work. It appears that's all it waqs. > > Without the power connecter the card will probe, and even > appear to be working but when the lines ri...

I've started to implement checks for package versions on dependencies in install.packages(). However, this is revealing a number of problems/misconceptions. (A) We do not check versions when loading namespaces, ahd the namespace registry does not contain version information. So that for example (rtracklayer) Depends: R (>= 2.7.0), Biobase, methods, RCurl Imports: XML (>= 1.98-0), IRanges, Biostrings will never check the version of namespace XML that is loaded, either already loaded or resulting fr...

Hello, As i check the call logs, some of my clients seem to make successful calls but, in logfiles, Source field seems empty..Still I can see who is the source from Channel tab as SIP/XXXX, and the called number and the time etc but.. nothing on Source and the Called ID tab. Just some clients has this problem. But as i check nothing special in their settings. What might cause this problem. Using

Hi, I use realtime and I found that changing accountcode needed to restart asterisk to activate that code and shown in CDR. Does it has a way to update accountcode without restart asterisk? ango

Hi All. Can anyone tell what is sip tunneling ahd how its works want a make a sip tunnel between to Asterisk Server.... Please share your Kind Toughts -- Best Regards Shazi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090820/091e1ed5/attachment-0001...

...ll count supported by this hardware? 2- do you suggest using 64bit Centos or other OS? 3- for such usage what codecs do you prefere? 711u? 723? gsm? knowing that we are after good quality calls. my experiences are with small call centers up to 40 seats .. Thank you for your help and support. -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 _________________________________________________________________ Hotmail: Trusted email with Microsoft?s powerful SPAM protection. http://clk.atdmt.com/GBL/go/201469226/di...

...ve so I was there early But I dropped a 4l erlenmeyer and slipped and fell on the broken glass. They had to take me to er and the attending physicians speculated if Iwould make it through the night. One thing led to another and I woulnd up In jail chained to a man named Daisy. After my injury, I ahd a grand mal siezure. After that my gun-toting privelgaes had which revoked because I was In a psych ward. I wish they would have told me. One thing led to another and I woulnd up In jail chained to a man named Daisy. My injury and legal problems are behin me now. I am ready for work-Wash U or ~...

...ng prototype and script.aculo.us? I think we need to: 1. get X and Y position of the Draggable 2 set revert = true or false by calculating positiong of the nearest Droppable 3 If Draggable is accepted by Droppable - set snap:[xd,yd], where xd and yd - Droppable coordinates But I am beginner in JS ahd aculo.us and cant write functions for REVERT and SNAP Help me please to do script like this by using aculo.us Thanks --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the Google Groups "Ruby on Rails: Spinoffs" group. To post...

..., there are good suggestions. > > regards > > On 1/20/06, Michakl Gaudette <michael.gaudette@virtutel.ca> wrote: > >>Hello, >> >>I'm a bit new to SIP, and I've set up a SIP line with Asterisk and my >>wholesale provider. That worked, fine. I ahd to open up the ports on my >>router, forward them to the correct box, again fine. >> >>Now, if I get one of my customers to connect his SIP phone to my Asterisk >>box, and HE'S behind a NAT firewall, does he have to go through the same >>process, or is it just the...

Is it possible to make use of queues for incoming calls but to have agents that do not need to log in ? If I create a queue and make certain SIP-users member of the queue, do these SIP-users always need to log in to the queue to be able to receive calls that are in the queue ?? Can't a member be just available when the phone is registered to the Asterisk-server ? In stead of also having to

Hi I have On func_odbc [EXEC] readhandle=ressqlserver writehandle=ressqlserver readsql=${ARG1} writesql=${ARG1} I'm trying an update on dialplan: exten=> 141,3,Set(dummy=${ODBC_EXEC(UPDATE Tabla set campo = ${EXTEN})}) On Cli: WARNING[3579]: func_odbc.c:353 acf_odbc_read: Error -1 in FETCH [UPDATE Tabla set campo = 4356] Any idea why is this?? The query

...problem, because in /var/log/messages it reports "Aug 21 17:41 peagasus ((which is my linux box)) PAM_pwdb [4277]: authentication failure; (uid=0) -> ben$. The password is correct, both in the database when I use "smbpasswd -a username" and when I type it. I thought encryption ahd something to do with it so I changed allowing null passwords. It didn't work either. Any suggestions?

I am looking for Video Conference Software (Open Source) , But but not for free Trial.. please give reference about it. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090427/cc9690ee/attachment.htm